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1 // Copyright 2014 The Chromium Authors. All rights reserved. | 1 // Copyright 2014 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ | 5 #ifndef MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ |
6 #define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ | 6 #define MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ |
7 | 7 |
8 #include <stddef.h> | 8 #include <stddef.h> |
9 #include <stdint.h> | 9 #include <stdint.h> |
10 | 10 |
11 #include <string> | 11 #include <string> |
12 | 12 |
13 #include "base/callback.h" | 13 #include "base/callback.h" |
14 #include "base/stl_util.h" | 14 #include "base/stl_util.h" |
| 15 #include "media/cast/common/rtp_time.h" |
15 #include "media/cast/net/cast_transport_defines.h" | 16 #include "media/cast/net/cast_transport_defines.h" |
16 | 17 |
17 namespace media { | 18 namespace media { |
18 namespace cast { | 19 namespace cast { |
19 | 20 |
20 enum Codec { | 21 enum Codec { |
21 CODEC_UNKNOWN, | 22 CODEC_UNKNOWN, |
22 CODEC_AUDIO_OPUS, | 23 CODEC_AUDIO_OPUS, |
23 CODEC_AUDIO_PCM16, | 24 CODEC_AUDIO_PCM16, |
24 CODEC_AUDIO_AAC, | 25 CODEC_AUDIO_AAC, |
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95 // The label associated with the frame upon which this frame depends. If | 96 // The label associated with the frame upon which this frame depends. If |
96 // this frame does not require any other frame in order to become decodable | 97 // this frame does not require any other frame in order to become decodable |
97 // (e.g., key frames), |referenced_frame_id| must equal |frame_id|. | 98 // (e.g., key frames), |referenced_frame_id| must equal |frame_id|. |
98 uint32_t referenced_frame_id; | 99 uint32_t referenced_frame_id; |
99 | 100 |
100 // The stream timestamp, on the timeline of the signal data. For example, RTP | 101 // The stream timestamp, on the timeline of the signal data. For example, RTP |
101 // timestamps for audio are usually defined as the total number of audio | 102 // timestamps for audio are usually defined as the total number of audio |
102 // samples encoded in all prior frames. A playback system uses this value to | 103 // samples encoded in all prior frames. A playback system uses this value to |
103 // detect gaps in the stream, and otherwise stretch the signal to match | 104 // detect gaps in the stream, and otherwise stretch the signal to match |
104 // playout targets. | 105 // playout targets. |
105 uint32_t rtp_timestamp; | 106 RtpTimeTicks rtp_timestamp; |
106 | 107 |
107 // The common reference clock timestamp for this frame. This value originates | 108 // The common reference clock timestamp for this frame. This value originates |
108 // from a sender and is used to provide lip synchronization between streams in | 109 // from a sender and is used to provide lip synchronization between streams in |
109 // a receiver. Thus, in the sender context, this is set to the time at which | 110 // a receiver. Thus, in the sender context, this is set to the time at which |
110 // the frame was captured/recorded. In the receiver context, this is set to | 111 // the frame was captured/recorded. In the receiver context, this is set to |
111 // the target playout time. Over a sequence of frames, this time value is | 112 // the target playout time. Over a sequence of frames, this time value is |
112 // expected to drift with respect to the elapsed time implied by the RTP | 113 // expected to drift with respect to the elapsed time implied by the RTP |
113 // timestamps; and it may not necessarily increment with precise regularity. | 114 // timestamps; and it may not necessarily increment with precise regularity. |
114 base::TimeTicks reference_time; | 115 base::TimeTicks reference_time; |
115 | 116 |
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140 virtual ~PacketSender() {} | 141 virtual ~PacketSender() {} |
141 }; | 142 }; |
142 | 143 |
143 struct RtcpSenderInfo { | 144 struct RtcpSenderInfo { |
144 RtcpSenderInfo(); | 145 RtcpSenderInfo(); |
145 ~RtcpSenderInfo(); | 146 ~RtcpSenderInfo(); |
146 // First three members are used for lipsync. | 147 // First three members are used for lipsync. |
147 // First two members are used for rtt. | 148 // First two members are used for rtt. |
148 uint32_t ntp_seconds; | 149 uint32_t ntp_seconds; |
149 uint32_t ntp_fraction; | 150 uint32_t ntp_fraction; |
150 uint32_t rtp_timestamp; | 151 RtpTimeTicks rtp_timestamp; |
151 uint32_t send_packet_count; | 152 uint32_t send_packet_count; |
152 size_t send_octet_count; | 153 size_t send_octet_count; |
153 }; | 154 }; |
154 | 155 |
155 struct RtcpReportBlock { | 156 struct RtcpReportBlock { |
156 RtcpReportBlock(); | 157 RtcpReportBlock(); |
157 ~RtcpReportBlock(); | 158 ~RtcpReportBlock(); |
158 uint32_t remote_ssrc; // SSRC of sender of this report. | 159 uint32_t remote_ssrc; // SSRC of sender of this report. |
159 uint32_t media_ssrc; // SSRC of the RTP packet sender. | 160 uint32_t media_ssrc; // SSRC of the RTP packet sender. |
160 uint8_t fraction_lost; | 161 uint8_t fraction_lost; |
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177 lhs.ntp_fraction == rhs.ntp_fraction && | 178 lhs.ntp_fraction == rhs.ntp_fraction && |
178 lhs.rtp_timestamp == rhs.rtp_timestamp && | 179 lhs.rtp_timestamp == rhs.rtp_timestamp && |
179 lhs.send_packet_count == rhs.send_packet_count && | 180 lhs.send_packet_count == rhs.send_packet_count && |
180 lhs.send_octet_count == rhs.send_octet_count; | 181 lhs.send_octet_count == rhs.send_octet_count; |
181 } | 182 } |
182 | 183 |
183 } // namespace cast | 184 } // namespace cast |
184 } // namespace media | 185 } // namespace media |
185 | 186 |
186 #endif // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ | 187 #endif // MEDIA_CAST_NET_CAST_TRANSPORT_CONFIG_H_ |
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